Re: [Freeswitch-users] Polycom phone registration problem

17 Nov 2011

      Couldn't you do something like 

if $contact.address != ip.src_addr && sip_agent contains Polycom THEN NDLB-connectile-dysfunction 

-Matt 

----- Original Message -----
...
From: "Michael Collins" <msc@freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org>
Sent: Thursday, November 17, 2011 2:48:32 PM
Subject: Re: [Freeswitch-users] Polycom phone registration problem
...
Hi Elliot,
...
IIRC, the scenario where the Polycom is behind NAT and the FS is on a
public IP is one where Polycom fails because they are too stubborn
to add rport to their SIP handling. (We've been barking at them for
over 5 years on this issue.)
...
There is a workaround in FS but it breaks non-Polycom phones that are
on the same profile, so you'll need to have a separate profile for
your Polycoms if you are in a mixed environment. In any case, for
the sofia profile that has the Polycoms you should check out these
two parameters:
...
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#NDLB-force-rport
http://wiki.freeswitch.org/wiki/NAT_Traversal#NDLB-connectile-dysfunction
...
You might also want to hop into #freeswitch on irc.freenode.net and
talk to some of our resident Polycom veterans.
...
-MC
...
On Wed, Nov 16, 2011 at 9:14 PM, Elliott Vogel <
elliott@zoogmedia.com > wrote:
...
...
Hello, I’m trying to setup freeswitch and I’m having a problem with
a
polycom phone registering. The phone is behind nat but the
freeswitch is on a public network can anyone tell me if I don’t
have
something configured correctly or where should look for the
problem?

...
...
< include >
...
< domain name = " zoogmedia.com " >
...
< params >
...
< param name = " dial-string " value = "
{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}
" />
...
</ params >

...
...
< variables >
...
< variable name = " record_stereo " value = " true " />
...
< variable name = " default_gateway " value = "
$${default_provider}
" />
...
< variable name = " default_areacode " value = " 312 " />
...
< variable name = " transfer_fallback_extension " value = "
operator
" />
...
</ variables >

...
...
< groups >
...
< group name = " default " >
...
< users >
...
< user id = " 14149823263 " >
...
< params >
...
< param name = " password " value = " 123456 " />
...
</ params >
...
< variables >
...
< variable name = " sip-force-contact " value = "
NDLB-connectile-dysfunction " />
...
< variable name = " toll_allow " value = "
domestic,international,local " />
...
< variable name = " accountcode " value = " 1000 " />
...
< variable name = " user_context " value = " default " />
...
< variable name = " outbound_caller_id_name " value = " Sales " />
...
< variable name = " outbound_caller_id_number " value = "
+14149823263 " />
...
</ variables >
...
</ user >
...
</ users >
...
</ group >
...
</ groups >

...
...
</ domain >
...
</ include >

...
...
< configuration name = " sofia.conf " description = " sofia
Endpoint
" >

...
...
< global_settings >
...
< param name = " log-level " value = " 0 " />
...
<!-- <param name="auto-restart" value="false"/> -->
...
< param name = " debug-presence " value = " 0 " />
...
<!-- <param name="capture-server" value="udp: homer.domain.com:5060
"/> -->
...
</ global_settings >

...
...
< profiles >

...
...
< profile name = " endpoints " >
...
< gateways >
...
< gateway name = " TNG " >
...
< param name = " username " value = " user " />
...
< param name = " password " value = " password " />
...
< param name = " proxy " value = " 69.25.128.195:5060 " />
...
< param name = " from-domain " value = " $${local_ip_v4}:5060 " />
...
< param name = " dtmf-type " value = " rfc2833 " />
...
< param name = " extension-in-contact " value = " true " />
...
< param name = " caller-id-in-from " value = " true " />
...
< param name = " register " value = " false " />
...
</ gateway >
...
</ gateways >
...
< aliases >

...
...
</ aliases >
...
< domains >
...
< domain name = " all " alias = " false " parse = " true " />
...
</ domains >
...
< settings >
...
< param name = " debug " value = " 0 " />
...
< param name = " sip-trace " value = " no " />
...
< param name = " sip-capture " value = " no " />
...
< param name = " rtp-ip " value = " $${external_rtp_ip} " />
...
< param name = " sip-ip " value = " $${external_sip_ip} " />
...
< param name = " ext-rtp-ip " value = " $${external_rtp_ip} " />
...
< param name = " ext-sip-ip " value = " $${external_sip_ip} " />
...
< param name = " sip-port " value = " 5070 " />
...
< param name = " tls " value = " $${external_ssl_enable} " />
...
< param name = " tls-bind-params " value = " transport=tls " />
...
< param name = " tls-sip-port " value = " $${external_tls_port} "
/>
...
< param name = " tls-cert-dir " value = " $${external_ssl_dir} " />
...
< param name = " tls-version " value = " $${sip_tls_version} " />

...
...
< param name = " rfc2833-pt " value = " 101 " />
...
< param name = " dialplan " value = " XML " />
...
< param name = " context " value = " default " />
...
< param name = " dtmf-duration " value = " 2000 " />
...
< param name = " inbound-codec-prefs " value = "
$${global_codec_prefs} " />
...
< param name = " outbound-codec-prefs " value = "
$${outbound_codec_prefs} " />
...
< param name = " hold-music " value = " $${hold_music} " />
...
< param name = " rtp-timer-name " value = " soft " />
...
< param name = " local-network-acl " value = " localnet.auto " />
...
< param name = " manage-presence " value = " false " />
...
< param name = " inbound-codec-negotiation " value = " greedy " />
...
< param name = " nonce-ttl " value = " 60 " />
...
< param name = " auth-calls " value = " false " />
...
< param name = " rtp-timeout-sec " value = " 300 " />
...
< param name = " rtp-hold-timeout-sec " value = " 1800 " />
...
< param name = " rtp-rewrite-timestamps " value = " true " />

...
...
< param name = " track-calls " value = " true " />
...
</ settings >
...
</ profile >

...
...
</ profiles >

...
...
</ configuration >

...
...
<? xml version = " 1.0 " encoding = " utf-8 " ?>
...
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
...
< include >
...
< context name = " default " >
...
< extension name = " unloop " >
...
< condition field = " ${unroll_loops} " expression = " ^true$ " />
...
< condition field = " ${sip_looped_call} " expression = " ^true$ "
...
...
< action application = " deflect " data = " ${destination_number} "
/>
...
</ condition >
...
</ extension >
...
< extension name = " check_auth " continue = " true " >
...
< condition field = " ${sip_authorized} " expression = " ^true$ "
break = " never " >
...
< anti-action application = " respond " data = " 407 " />
...
</ condition >
...
</ extension >
...
< extension name = " local " >
...
< condition field = " ${toll_allow} " expression = " local " />
...
< condition field = " destination_number " expression = "
^([0-9]{7})$ " >
...
< action application = " set " data = "
effective_caller_id_name=${outbound_caller_id_name} " />
...
< action application = " set " data = "
effective_caller_id_number=${outbound_caller_id_number} " />
...
< action application = " bridge " data = "
sofia/gateway/TNG/+1${default_areacode}$1 " />
...
</ condition >
...
</ extension >
...
< extension name = " domestic " >
...
< condition field = " ${toll_allow} " expression = " domestic " />
...
< condition field = " destination_number " expression = "
^(1{0,1}\d{10})$ " >
...
< action application = " set " data = "
effective_caller_id_name=${outbound_caller_id_name} " />
...
< action application = " set " data = "
effective_caller_id_number=${outbound_caller_id_number} " />
...
< action application = " bridge " data = " sofia/gateway/TNG/+$1 "
/>
...
</ condition >
...
</ extension >
...
< extension name = " international " >
...
< condition field = " ${toll_allow} " expression = " international
"
/>
...
< condition field = " destination_number " expression = "
^011?(\d+)$
" >
...
< action application = " set " data = "
effective_caller_id_name=${outbound_caller_id_name} " />
...
< action application = " set " data = "
effective_caller_id_number=${outbound_caller_id_number} " />
...
< action application = " bridge " data = " sofia/gateway/TNG/+$1 "
/>
...
</ condition >
...
</ extension >
...
</ context >
...
</ include >

...
...
_________________________________________________________________________
...
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...
consulting@freeswitch.org
...
http://www.freeswitchsolutions.com

...
...
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...
http://www.cudatel.com

...
...
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...
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...
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...
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
...
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
...
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
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Re: [Freeswitch-users] Polycom phone registration problem

Matthew S. Crocker